Sorry for that, may be you have some experience about why conference button not working? every thing is fine calls, transfer, etc. problems only with conference
Hi
Thanks for your work and support.
Iām trying to get my cisco 9971 to register with asterisk (11.3.0)+freePBX (2.11.0.0beta2.8) running on Raspberry Pi (http://www.raspberry-asterisk.org/)
my question here is: How to patch my (installed and running) asterisk on R.Pi with the files on:
https://issues.asterisk.org/jira/browse/13996
Iāve figured (yet Iām not sure) that I only need to apply the āgareth-11.3.0.patchā patch. I donāt know if that is all I need.
There are also two patches called: video2-new.patch and smime.p7s on this website linked at:
http://lists.digium.com/pipermail/asterisk-video/2011-October/003560.html
but Iām not sure if they are already included in the latest gareth-11.3.0.patch or how to know for sure if they are or are not.
If they are not then Iāll need to also apply them. I donāt know if should I apply them before or after gareth-11.3.0.patch
but it seems to me that those 2 patches (video2-new.patch and smime.p7s) are for asterisk-1.8.4.4!! I hope gareth-11.3.0.patch is enough.
Iām trying to follow your explanation on how to configure 9971 cisco phone to register with asterisk but till now was not successful. I have yet to see a registration attempt from asterisk Cli.
on the phone web log I found some where that it tried to register sip (correct sip to correct asterisk ip) but did not get a respond.
my cisco 9971 FW is sip9971.9-2-3-27. I have the update to 9.3.1.33 but Iāve figured it is better to wait to make sure before updating just in case the older firmware was more reliable/asterisk friendly as i donāt know how to downgrade the FW after updating.
I think I have got the phone configuration file right and only added 2 sip extensions to be utilized in the phone and configured it in the SEP(mac_address).cnf.xml file but still phone is not registering. so I guess there is no more path for me to go except patching asterisk and to hope that this is all I need to get it to work.
another question about the NTP
is it:
D/M/Ya
or
D/M/YA
or
D/M/Y
as I still didnāt get my clock to update with NTP: 203.88.112.222
more details about my settings/configurations will be further gladly provided if needed.
thanks for your help.
Did you ever solve this,
I note my phones seem to be ignoring the
video_imageattr=recv [x=640,y=480,q=0.50]
and using 176x144
Iām willing to even pay someone for some support to get my PBX up and running, I think I have the configuration file built correctly but my phone keeps saying it canāt download configuration file from 192.xxx.x.xx . Iām still a linux noob in a way so any help would be appreciated and if I need to pay for support to learn this I will, this is more or less just going to be for personal gain / use, Iām a nerd I love to play with technology but there are times it gets a upper hand on me
So got my phone today and trying for the life of me to get it work but I get the usual Error Verifying Config Info, Iāve pretty much taken the sample in this post and tried even just doing a copy and paste with getting this same error. I then tried to remove some extra items that I wouldnāt use like speed buttons etc and same issue,
I cant for the life of me get the code to function correctly on here so Iāve put it in a pastebin
Well scrap my previous question got the phone to register, so the only two features I canāt figure out still are how to get to missed calls / contact book. That pastbin config is no longer valid I modified it a few times and got it to work.
So question 1) When I hit the address book button next to the settings gear it says service not available, so how would I get it to show local services (Address Book, Missed Calls, Placed Calls)
and 2) does conference calling on the phone work yet?
Please post your new working config itās something very popular.
I donāt think the conferencing has been fixed, I donāt know I still canāt get mine to register so I am looking forward to your post.
The Cisco directory is a web page, do you know how to add content to httpd (apache) web server? If so Google Cisco Phone Apps, there are many free scripts out on the net.
I can get my phone to register now as well, placed a call with my google line and was able to call out, having problems getting my incoming calls on google to work though, I had it working once but have been unable to get it to work since I reinstalled. I tried using the forwarding script in here to but it seems theres a bug with it, but so far Iām figuring out most of the stuff, contacts Iām still working on as well, and a few others, Iām guessing there in the features.conf but I have yet to locate an example one that I can go off of.
So are either of you guys going to share a working config?
Many thanks for posting this tutorial. I got my Cisco 8961 working properly.
The only part I still have issues is WMI. I use voip.ms as provider and for some reason there is no notification on my device for new messages.
[general]
ā¦
mwi => 123456:[email protected]:5060/101
Just to note I couldnāt get my phone to download from the tftpboot folder on my freepbx server, I ended up downloading a free open source tftp server for my windows 7 box and transferring firmware/config files to my phone that way.
Here is my main config file labled SEPMACADDRESSOFPHONE.cnf.xml Note this has to be all caps or the phone wonāt take it.
<?xml version="1.0" encoding="UTF-8"?> SIPadmin
cisco
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>IP ADDRESS OF YOUR PBX</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
true
3
SIP8941_8945.9-3-2-12
DefaultFP.xml
United_States
<networkLocaleInfo>
<name>English_United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
1
0
96
0
96
2
5
1
0
0
1
false
0
0
0
3804
false
USECALLMANAGER
5060
USECALLMANAGER
5060
true
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>true</retainForwardInformation>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>0</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>true</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress>IP ADDRESS OF YOUR PBX</natAddress>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<dscpForTelepresence>128</dscpForTelepresence>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<softKeyFile>softkey.xml</softKeyFile>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>This can be anything it seems</phoneLabel>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>Line Label on phone</featureLabel>
<name>FreePBX Extension you setup</name>
<displayName>Incoming Label I think?</displayName>
<contact> FreePBX Extension you setup </contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName> FreePBX Extension you setup </authName>
<authPassword>FreePBX Extension Secret/Password</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>Voicemail Extension go's here</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>4</maxNumCalls>
<busyTrigger>2</busyTrigger>
</line>
</sipLines>
0
Missed Calls
Application:Cisco/MissedCalls
Voicemail
Application:Cisco/Voicemail
Received Calls
Application:Cisco/ReceivedCalls
Placed Calls
Application:Cisco/PlacedCalls
Personal Directory
Application:Cisco/PersonalDirectory
Corporate Directory (Demo)
http://directory.ciscoxmlservices.com/demo1/demo1/
Next is my dialplan.xml
Heres my XMLDefault.cnf.xml
2000 5060 5061 FreePBX IP Firmware Version see belowMy firmware was SIP8941_8945.9-3-2-12 now also to note
I had 10 files labeled like SIP8941_8945.9-3-2-12.bin1.sgn except bin1 would be like bin2 etc, then there was a .loads after one of them. Another thing to note is there are multiple firmware files on Ciscoās site, the ones that have a SP at the end or wherever are service packs, they do not include the BOOT8941_8945.0-0-1-0.bin.sgn file which you need in order to initiate the firmware upgrade, I would recommend doing the older firmware first then the service pack. If I can help out anymore let me know, Iām just slow to getting back to my posts sorry about that.
So Iāve been staring at this for several hours a day over the weekend and Iām missing something. I set up a new box in the lab, CentOS 6, Asterisk 11.5.1, FreePBX 2.11, chan_sccp 4 (for 7921), and loaded Garethās presence patch from JIRA. No matter what I do I canāt get these phones to register over TCP for presence to work.
In the FreePBX GUI I have transport set to All - TCP Primary. If I set TCP only the phones wont register at all. Allowing UDP lets registrations and calls occur, but presence is ignored. Whatās also strange is the configs that were shared here list UDP as the transport. This line in the SEP.XML defines UDP as the transport:
2
Iāve tried using 1 and 4 here with no avail. Packet captures on the pbx show the phone doesnāt even make an attempt on port 5060. In UDP mode I do see the presence notification and acknowledgment. (Note, I mean SIP acknowledgement not layer 4 TCP ack flag).
22:14:06.618205 IP 10.0.10.55.sip > 10.0.5.101.sip: SIP, length: 855
E`.s....@...
.
7
..e....._'.NOTIFY sip:[email protected]:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 72.78.211.55:5060;branch=z9hG4bK746828ea
Max-Forwards: 70
From: ;tag=as2ec9a175
To: ;tag=e840400d3bcc00043b34d142-171f7a7c
Contact:
Call-ID: [email protected]
CSeq: 107 NOTIFY
User-Agent: FPBX-2.11.0(11.5.1)
Subscription-State: active;expires=3600
Event: presence
Content-Type: application/cpim-pidf+xml
Content-Length: 345
<?xml version="1.0"?>
22:14:06.641256 IP 10.0.5.101.49159 > 10.0.10.55.sip: SIP, length: 416
E`ā¦>ā¦P
ā¦e
.
7ā¦SIP/2.0 200 OK
Via: SIP/2.0/UDP 72.78.211.55:5060;branch=z9hG4bK746828ea;received=10.0.10.55
From: sip:[email protected];tag=as2ec9a175
To: sip:[email protected];tag=e840400d3bcc00043b34d142-171f7a7c
Call-ID: [email protected]
Date: Mon, 09 Sep 2013 02:14:06 GMT
CSeq: 107 NOTIFY
Server: Cisco-CP9971/9.3.4
Contact: sip:[email protected]:5060;transport=UDP
Content-Length: 0
I did set ciscounified=yes under the extensions in sip_custom_post.conf as well. Here is my XML file for the phone. Iāve left everything intact as this is in a lab with no access to anything real.
SIP
M/D/YA
Eastern Standard/Daylight Time
0.0.0.0
unicast
5060
10.0.10.55
true
true
true
service-uri-cfwdall
service-uri-pickup
service-uri-gpickup
2
true
2
2
2
true
true
9
3083
USECALLMANAGER
5060
3083
2
1
3083
Passw0rd123
false
1111
5
true
true
true
true
9
Intercom
USECALLMANAGER
5060
13083
3
1
13083
Passw0rd123
false
1111
5
true
true
true
true
21
Mr. Burns
3081
1
21
Smithers
3802
1
21
Service Queue
*45
1
false
g711ulaw
dialplan.xml
softkeyDefault.xml
true
true
1
0
H. Simpson
2
true
184
136
admin
2
featurePolicyDefault.xml
sip9971.9-3-4-24.loads
1
1
admin
Lab123
2
Missed Calls
Application:Cisco/MissedCalls
Received Calls
Application:Cisco/ReceivedCalls
Placed Calls
Application:Cisco/PlacedCalls
Voicemail
Application:Cisco/Voicemail
0003298300203293
English_United_States
en
United_States
United_States
http://phone-xml.berbee.com/menu.xml
0
96
2
Hello,
Does anybody know the config file parameters needed to enable use of a KEM with the 9971 phone?
I am using a couple of these phones at home and have everything functional, with loads of help from this and other such threads I found in the community. Unfortunately, Iāve not been able to discover anything about configuring or enabling a KEM.
I got a KEM for the 9971 phone and when I plug it in the phone shows a message that the LKEM has been disabled by the administrator. I found CISCO documentation describing how to enable it via Call Manager and I presume that would create some new entries in the config file for the phone. Any thoughts on how I might get a hold of those settings?
Have you just tried programming key11? I have a 9971 at home. I may just pick up the sidecar for fun.
Someone mentioned the missing dial key to me the other day. I had never seen it. Moved a 9951 from one server to the another and now the dial key is missing (actually all the feature keys) and the camera says disabled by administrator.
Very strange, no config changes made.
I tried changing the camera variables from 1 to 0 just to see a change and there was none.
If anyone know the cause of these symptoms let me know. I will keep hacking away while I watch the Cleveland Browns lose another game.
I tried programming several extra keys - 7, 8, 30, 22, etc., randomly to see if something helps. Not sure of the significance of 11 since the phone has 6 keys but I tried the 11 also now. The config file is accepted by the phone without any problems but the KEM still doesnāt work.
CKEM
36
Unfortunately the phone cannot power up the KEM even now - seems not to have enough power coming from a cheap AT POE injector I was using. Have ordered an "expensive" one to see if that does the trick
@simcity I got this 9971 on my desk a year ago - i followed your step that time and had it registered. Now though had to relocate and started from scratch. I am kind of spending days and night with no luck in registering the phone. Keeps saying: āError Verifying Config Infoā although the SEPā¦cnf.xml is a copy of yours above! Help please⦠Ciao
@simcity I got this 9971 on my desk a year ago - i followed your step that time and had it registered. Now though had to relocate and started from scratch. I am kind of spending days and night with no luck in registering the phone. Keeps saying: āError Verifying Config Infoā although the SEPā¦cnf.xml is a copy of yours above! Help please⦠Ciao
Let me express my gratitude for sharing this information.
Iām currently on a quest to connect
3 cisco9951
2 cisco9971 + cams
2 2N Helios
to an AVAYA IPOFFICE 500 V2.
The information provided was paramount in resurrecting the "zombified"
cisco terminals.
Thank you !
Merry Christmas